Re: [Asterisk-Users] Help Connecting Cisco AS5300 to Asterisk

[Stefan Reuter]

 It doesn't have to be IAX. Do you know how to
 configure it with another protocol?

have a look at
http://ertw.com/blog/archives/asterisk_and_an_as5350_sip_peer-190405.html

=Stefan

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Re: [Asterisk-Users] FXO Gateway recommendation

[Wai-Sun Chia]

On 6/8/05, VoIP Newbie [EMAIL PROTECTED] wrote:
 My 4-port FXO is only $300.
 
Which product/model are you using then?

/wai-sun
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RE: [Asterisk-Users] MGCP Useragent

[Florian Overkamp]

Hi, 

 -Original Message-
 1- Anybody implement mgcp useragent in *.

Nope. Hasn't been done yet.

 2- Where can i get that.

Not available in your nearest drugstore.

 3- if no then anybody can help me to write it down.

Digium ?

Florian


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RE: [Asterisk-Users] error message: INIT: Id s0 respawning too fast:disable for 5 minutes

[Florian Overkamp]

Hi, 

 -Original Message-
 I have set up [EMAIL PROTECTED] with Digium TDM400P 2FXO/2FXS.
 I am unable to seize my trunks from either soft or analog phones.
 Inbound calls result in answer/disconnection.
  
 I see the following error code on my asterisk server
  
 INIT: Id s0 respawning too fast: disable for 5 minutes
  
 Does anyone have any suggestions for me?
 I'd really appreciate some help on this.

I don't think that's asterisk related at all. Check /etc/inittab to see what
's0' is. My guess is it's a serial port that you have connected to something
proprietary.

If that is the case, just comment out the s0 line and 'telinit q' should
stop the messages.

Florian


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RE: [Asterisk-Users] gxp-2000 tftp cfg

[Peter Svensson]

On Wed, 8 Jun 2005, David Phelan wrote:

  If you download the configuration tool which I couldn't get working on my
 systemthere is a cfg template in there for 1.0.1.8

Oh, then they have added it, or we missed it the first time around. We 
have it running. We had to tweak the paths in the file encode.sh a bit 
to match our setup. 

Peter


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[Asterisk-Users] Asterisk to Avaya PBX using TDM cards

[Billy]

Hi 
 I'm new in this field, have been reading a lot, and have a little question. 
could it be possible to connect an Avaya IP office pbx to asterisk using a 
E1/T1/Pri?

Original instalation:

Telefone company|Pri---Pri|IP Pffice

My Question:

Telefone company|Pri ---TDM|Asterisk|TDM ---Pri|IP Office

I know that it can be done by using h323, but I need a card on the IPOffice my 
problem is that I have no more room for expantion on this pbx so I was 
thinking instead of upgrading the IPOffice maybe I can start using *. 
The secret of success is converting your problems into opportunities

Thanks to everyone.

Billy

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[Asterisk-Users] Newbie on asterisk ask for configuratio help

[craz sead]

Hi all,

iam a student trying to build an asterisk pbx as a
simple configuration only two extention (using
Xlite)without outsite telephone line. i already follow
the instruction and seem the asterisk work fine
because there is no error message. when i configure
SIP.conf and extention.conf i hope the phone will ring
each other as an extention. but it doesn't work. i
follow the instruction at
www.automated.it/guidetoasterisk.htm this site but
nothing happen.

please help if someone have a configuration file or
any book that i can download and read because i am
really newbie here

thks

roywish 

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RE: [Asterisk-Users] error message: INIT: Id s0 respawning toofast:disable for 5 minutes

[Wagner Gimenes]

Guys (and Gals),

FYI I also have the *same* message here. Wonder is it is related to my
Compaq D500 Space Saver PIV 1.7 or the fact that I don't yet have a
modem card in the * box.

(Please don't shoot me, did try Google first)

Many thanks,

Wagner Gimenes 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Florian
Overkamp
Sent: 08 June 2005 08:16
To: [EMAIL PROTECTED]; 'Asterisk Users Mailing List -
Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] error message: INIT: Id s0 respawning
toofast:disable for 5 minutes

Hi, 

 -Original Message-
 I have set up [EMAIL PROTECTED] with Digium TDM400P 2FXO/2FXS.
 I am unable to seize my trunks from either soft or analog phones.
 Inbound calls result in answer/disconnection.
  
 I see the following error code on my asterisk server
  
 INIT: Id s0 respawning too fast: disable for 5 minutes
  
 Does anyone have any suggestions for me?
 I'd really appreciate some help on this.

I don't think that's asterisk related at all. Check /etc/inittab to see
what 's0' is. My guess is it's a serial port that you have connected to
something proprietary.

If that is the case, just comment out the s0 line and 'telinit q' should
stop the messages.

Florian


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[Asterisk-Users] bypass incoming ring..is it possible?

[stevanus]

Hi,

Is it possible to bypass incoming ring on asterisk so that incoming 
calls come to asterisk box will be directed straight into did?


Is anyone able to give me any clues or pinpoint me where I can get more 
information about it?


Thanks for your attention..

Best regards,

Stevanus
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[Asterisk-Users] Xlite not communicating with Asterisk

[Mohamed A. Gombolaty]

Dear All,

I have downloaded the xlite version 2.0 for windows and I made the
following conf in the xlite itself as the document suggested in order to
make it work with Asterisk but still it doesn't work as a matter of fact
when I tried to make a tcp dump I can see no packets going between the
windows client and the Asterisk server at all, here is the my conf on
the xlite itself:

in the MenuSystem SettingsSIP ProxyDeafult

Enabled: yes
Display Name:
Username:
Authorization User:
Password: 
Domain/Realm: mysip.server.com
SIP Proxy: 192.168.99.243
Outbound Proxy:
Use Outblound Proxy: Default
Send internal IP: Always
Register: Always
Direct Dial IP: NO
DIal Prefix:



my sip.conf for the device is as follow:

[881]
;Turn off silence suppression in X-Lite (Transmit Silence=YES)!
;Note that Xlite sends NAT keep-alive packets, so qualify=yes is not
needed
type=friend
secret=
callerid=Mohamed Mahmoud 881
host=dynamic
dtmfmode=inband
context=from-sip
canreinvite=no
disallow=all
allow=gsm



ofcourse I added in the context mentioned above the macro I use with all
my extensions.



--
Thx
MAG



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Re: [Asterisk-Users] DISA Help

[Wilson Pickett]

 when i try to dial a number it just dies. 
Meaning what?   Silence? Hangup? 

Does dialing voicemail on that same setup work? That would tell
whether it hears the DTMF.
Other wise, check the codec and dtmf mode, some combinations don't
work on some phones.
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Re: [Asterisk-Users] Xlite not communicating with Asterisk

[Wilson Pickett]

 Enabled: yes
 Display Name:
 Username:
 Authorization User:
 Password: 
 Domain/Realm: mysip.server.com

Is this your username:
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Re: [Asterisk-Users] bypass incoming ring..is it possible?

[Alexander Ilyushin]

You can first answer to call, and then provide playtones(ring) to caller.2005/6/8, stevanus [EMAIL PROTECTED]:
Hi,Is it possible to bypass incoming ring on asterisk so that incomingcalls come to asterisk box will be directed straight into did?
Is anyone able to give me any clues or pinpoint me where I can get moreinformation about it?Thanks for your attention..Best regards,Stevanus___
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Re: [Asterisk-Users] Help Connecting Cisco AS5300 to Asterisk

[Ritesh Jalan]

You can connect it through H.323


Thanks  Regards
Ritesh Jalan
Senior Engineer - Test  Audit
Net4India Ltd.
703 Bikaji Cama Bhawan
11 Bikaji Cama Place
New Delhi - 110029
Ph: +91-11-26160129 ext. 131
Cell : +91-9818616329
Web site: http://www.net4india.com

==
This message may contain confidential and/or privileged information. If you
are not the addressee or authorized to receive this for the addressee, you
must not use, copy, disclose or take any action based on this message or any
information herein. If you have received this message in error, please
advise the sender immediately by reply e-mail and delete this message. Thank
you for your cooperation.

==
- Original Message - 
From: chawki hammoud [EMAIL PROTECTED]
To: Ritesh Jalan [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion asterisk-users@lists.digium.com
Sent: Wednesday, June 08, 2005 10:34 AM
Subject: Re: [Asterisk-Users] Help Connecting Cisco AS5300 to Asterisk




 --- Ritesh Jalan [EMAIL PROTECTED] wrote:

  Cisco dosent use IAX protocol,

 It doesn't have to be IAX. Do you know how to
 configure it with another protocol?



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RE: [Asterisk-Users] Xlite not communicating with Asterisk

[Shahan Kalutanthri]

HI..!!

Is you windows PC  the Asterisk in the same LAN.



-Original Message-
From: Mohamed A. Gombolaty [mailto:[EMAIL PROTECTED] 
Sent: Wednesday, June 08, 2005 2:29 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Xlite not communicating with Asterisk


Dear All,

I have downloaded the xlite version 2.0 for windows and I made the following
conf in the xlite itself as the document suggested in order to make it work
with Asterisk but still it doesn't work as a matter of fact when I tried to
make a tcp dump I can see no packets going between the windows client and
the Asterisk server at all, here is the my conf on the xlite itself:

in the MenuSystem SettingsSIP ProxyDeafult

Enabled: yes
Display Name:
Username:
Authorization User:
Password: 
Domain/Realm: mysip.server.com
SIP Proxy: 192.168.99.243
Outbound Proxy:
Use Outblound Proxy: Default
Send internal IP: Always
Register: Always
Direct Dial IP: NO
DIal Prefix:



my sip.conf for the device is as follow:

[881]
;Turn off silence suppression in X-Lite (Transmit Silence=YES)! ;Note that
Xlite sends NAT keep-alive packets, so qualify=yes is not needed type=friend
secret= callerid=Mohamed Mahmoud 881 host=dynamic dtmfmode=inband
context=from-sip canreinvite=no disallow=all allow=gsm



ofcourse I added in the context mentioned above the macro I use with all my
extensions.



--
Thx
MAG



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Re: [Asterisk-Users] Help Connecting Cisco AS5300 to Asterisk

[Ritesh Jalan]

A prefix will be passed for authentication from Asterisk to cisco AS5300


Thanks  Regards
Ritesh Jalan
Senior Engineer - Test  Audit
Net4India Ltd.
703 Bikaji Cama Bhawan
11 Bikaji Cama Place
New Delhi - 110029
Ph: +91-11-26160129 ext. 131
Cell : +91-9818616329
Web site: http://www.net4india.com

==
This message may contain confidential and/or privileged information. If you
are not the addressee or authorized to receive this for the addressee, you
must not use, copy, disclose or take any action based on this message or any
information herein. If you have received this message in error, please
advise the sender immediately by reply e-mail and delete this message. Thank
you for your cooperation.

==
- Original Message - 
From: chawki hammoud [EMAIL PROTECTED]
To: Ritesh Jalan [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion asterisk-users@lists.digium.com
Sent: Wednesday, June 08, 2005 10:34 AM
Subject: Re: [Asterisk-Users] Help Connecting Cisco AS5300 to Asterisk




 --- Ritesh Jalan [EMAIL PROTECTED] wrote:

  Cisco dosent use IAX protocol,

 It doesn't have to be IAX. Do you know how to
 configure it with another protocol?



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 Get on-the-go sports scores, stock quotes, news and more. Check it out!
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Re: [Asterisk-Users] Help Connecting Cisco AS5300 to Asterisk

[Alexander Ilyushin]

You can connect it only via SIP2005/6/8, chawki hammoud [EMAIL PROTECTED]:
Hi:I have been Googling around for documents of how toconfigure aCisco AS5300 to connect to the PSTNthrough Asterisk, IAX channel.Please help me configuring Cisco and IAX or send mesome documentation referral.
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Re: [Asterisk-Users] Message Playback

[Alexander Ilyushin]

You must answer the call anyway. And then playback some message2005/6/8, Sahil Gupta [EMAIL PROTECTED]:
Hi,I'd like to know how I can playback a pre-recorded message to a user usingour system without answering the call.I want to do the above in the scenario where the user dials a number andthe number has been dialled incorrectly.
Regards,Sahil GuptaVoiceValley___Asterisk-Users mailing listAsterisk-Users@lists.digium.com
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Re: [Asterisk-Users] FXO Gateway recommendation

[VoIP Newbie]

Please visit www.broad-tel.com for details.

On 6/8/05, Wai-Sun Chia [EMAIL PROTECTED] wrote:
 On 6/8/05, VoIP Newbie [EMAIL PROTECTED] wrote:
  My 4-port FXO is only $300.
 
 Which product/model are you using then?
 
 /wai-sun

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Re: [Asterisk-Users] Xlite not communicating with Asterisk

[Zoa]

http://www.asteriskguru.com/tutorials/asterisk_voip_ipphone.html
http://www.asteriskguru.com/tutorials/xlite_softphone.html

Read these two tutorials and you should be fine.

Zoa



Wilson Pickett wrote:


Enabled: yes
Display Name:
Username:
Authorization User:
Password: 
Domain/Realm: mysip.server.com




Is this your username:
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[Asterisk-Users] file.c:1073 ast_waitstream_full: Wait failed (Interrupted system call)

[Mark Ackroyd]

Hi

I have a PHP agi-bin scripted called callhander.php and it’s setup to
answer anything that comes into the PBX,

In the script I am trying to the get the system to play a file called home
which I know works, as I can get the Play function to work from the
extensions.conf file.  However within the script the STREAM FILE function
give this message in a debug

Jun  6 15:37:07 WARNING[776]: file.c:1073 ast_waitstream_full: Wait failed
(Interrupted system call)

The say numbers commands work no problem. The OS is FreeBSD 5.4 anyone got
any idea how to fix this?

Mark


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Re: [Asterisk-Users] Xlite not communicating with Asterisk

[Mohamed A. Gombolaty]

Hi Shahan,
yes both are in the same LAN
Thx
MAG

Shahan Kalutanthri wrote:
HI..!!
Is you windows PC  the Asterisk in the same LAN.
-Original Message-
From: Mohamed A. Gombolaty [mailto:[EMAIL PROTECTED]]
Sent: Wednesday, June 08, 2005 2:29 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Xlite not communicating with Asterisk
Dear All,
I have downloaded the xlite version 2.0 for windows and I made the following
conf in the xlite itself as the document suggested in order to make
it work
with Asterisk but still it doesn't work as a matter of fact when I
tried to
make a tcp dump I can see no packets going between the windows client
and
the Asterisk server at all, here is the my conf on the xlite itself:
in the Menu>System Settings>SIP Proxy>Deafult
Enabled: yes
Display Name:
Username:
Authorization User:
Password: 
Domain/Realm: mysip.server.com
SIP Proxy: 192.168.99.243
Outbound Proxy:
Use Outblound Proxy: Default
Send internal IP: Always
Register: Always
Direct Dial IP: NO
DIal Prefix:
my sip.conf for the device is as follow:
[881]
;Turn off silence suppression in X-Lite ("Transmit Silence"=YES)! ;Note
that
Xlite sends NAT keep-alive packets, so qualify=yes is not needed type=friend
secret= callerid="Mohamed Mahmoud" 881> host=dynamic dtmfmode=inband
context=from-sip canreinvite=no disallow=all allow=gsm
ofcourse I added in the context mentioned above the macro I use with
all my
extensions.
--
Thx
MAG
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--
Thx
MAG

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Re: [Asterisk-Users] Xlite not communicating with Asterisk

[Mohamed A. Gombolaty]

Hi Wilson,
yes I am leaving it blank although I did try to use a username in the
sip.conf but with the same result also I have tried to put the extension
881 but the same result.
Wilson Pickett wrote:
> Enabled: yes
> Display Name:
> Username:
> Authorization User:
> Password: 
> Domain/Realm: mysip.server.com
Is this your username:
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--
Thx
MAG

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[Asterisk-Users] no DTMF pass-thru

[Asterisk]

Hi all,We have a little problem.One of our customers has a problem with DTMF pass-thru.They use GrandStream 286 devices to connect their pstn phones to asterisk.everything works like a charm, except DTMF pass-thru. when they call an IVR system, they cannot select options because the DTMF tones never reach the IVR.Normal asterisk voicemail works though. so it looks like asterisk is not forwarding the DTMF tones to the zap interface.Is this a setup problem or asterisk intended?Regards.Andre

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RE: [Asterisk-Users] Xlite not communicating with Asterisk

[Shahan Kalutanthri]

Title: Message



on the 
asteriskconsole puta "sip debug" and see if you get any debug 
information.
coz 
even though you extension.conf or sip.conf is not properly configured still you 
should get the debug info..!!

shahan

  
  -Original Message-From: Mohamed A. 
  Gombolaty [mailto:[EMAIL PROTECTED] Sent: Wednesday, June 
  08, 2005 3:11 PMTo: Asterisk Users Mailing List - Non-Commercial 
  DiscussionSubject: Re: [Asterisk-Users] Xlite not communicating 
  with AsteriskHi Shahan, 
  yes both are in the same LAN 
  Thx MAG  
  Shahan Kalutanthri wrote: 
  HI..!! 
Is you windows PC  the Asterisk in the same LAN. 
-Original Message- From: Mohamed A. Gombolaty [mailto:[EMAIL PROTECTED]] 
Sent: Wednesday, June 08, 2005 2:29 PM To: 
asterisk-users@lists.digium.com Subject: [Asterisk-Users] Xlite not 
communicating with Asterisk 
Dear All, 
I have downloaded the xlite version 2.0 for windows and I made the 
following conf in the xlite itself as the document suggested in order to 
make it work with Asterisk but still it doesn't work as a matter of fact 
when I tried to make a tcp dump I can see no packets going between the 
windows client and the Asterisk server at all, here is the my conf on 
the xlite itself: 
in the MenuSystem SettingsSIP ProxyDeafult 
Enabled: yes Display Name: Username: Authorization User: 
Password:  Domain/Realm: mysip.server.com SIP Proxy: 
192.168.99.243 Outbound Proxy: Use Outblound Proxy: Default Send 
internal IP: Always Register: Always Direct Dial IP: NO DIal 
Prefix: 
my sip.conf for the device is as follow: 
[881] ;Turn off silence suppression in X-Lite ("Transmit 
Silence"=YES)! ;Note that Xlite sends NAT keep-alive packets, so 
qualify=yes is not needed type=friend secret= callerid="Mohamed 
Mahmoud" 881 host=dynamic dtmfmode=inband context=from-sip 
canreinvite=no disallow=all allow=gsm 
ofcourse I added in the context mentioned above the macro I use with all 
my extensions. 
-- Thx MAG 
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Thx
MAG 



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Re: [Asterisk-Users] Help Connecting Cisco AS5300 to Asterisk

[chawki hammoud]

I have already looked into this page.  I thought this
was for AS 5350, I am not familiar with Cisco products
and I don't know if there is a difference. And there
is no Asterisk set-up in this example.

Regards;

--- Stefan Reuter [EMAIL PROTECTED] wrote:

  It doesn't have to be IAX. Do you know how to
  configure it with another protocol?
 
 have a look at

http://ertw.com/blog/archives/asterisk_and_an_as5350_sip_peer-190405.html
 
 =Stefan
 
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Re: [Asterisk-Users] so what are the additional hardware componentsneeded?

[Steve Totaro]

keep reading

  - Original Message - 
  From: 
  infra 
  struct 
  To: asterisk-users@lists.digium.com 
  
  Sent: Tuesday, June 07, 2005 10:02 
  PM
  Subject: [Asterisk-Users] so what are the 
  additional hardware componentsneeded?
  
  I have 20 personal computers in LAN with full duplex soundcard, 
  microphone(headset) 
  I will use this setup for making PC to PC phone calls
  in addition I have a Linux server, in which i will be installing asterisk 
  and Internet Connection - DSL of speed 128kbps I will 
  be using asterisk for making calls to PSTN numbers(PC toPhone 
  calls)so what are the additional hardware components needed? 
  i figured out,Digium X100P (Asterisk, Linux server, X100P is for 
  PSTN connectivity (1 line) ) TDM400P on clients, in all Personal 
  Computers(which also have Softphones like SJPhone,As the softphones can 
  also make PSTN calls) 
  Please users any comments about my finding?
  
  
  Discover Yahoo!Get on-the-go sports scores, stock quotes, news  more. 
  Check 
  it out!
  
  

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[Asterisk-Users] performance of * in several scenarios

[barney]

Hi,

Is here someone who could provide meany information from 
practical using of * ?

I need to know more about performance. The 
main question is: 
"How many extensions should i have configuredin and 
provided with my * box in several cases":

1. * is usedonly for SIP signalling, no rtp stream is 
going through * (always using reinvite and no nat used in lan/wan)
2. * is used for signalling, rtp stream is going between UAs, 
but rtp stream is going through *, when is routed outside (from SIP to TDM 
world) via SIP trunk

3. * is used for signalling, rtp stream is going between UAs, 
but rtp stream is going through *, when is routed outside (from SIP to TDM 
world) via local CAPI/ZAP interface
4. * is used for signalling, rtp stream is always going 
through (never reinvite)

Common details:
- no codec translations / only one codec used in whole 
network 
- voicemail system and other services like wakeup calls, 
weather information, ... are running on other (dedicated) * box (hope that 
its possible)
- no frontend like SER used

Are there any tables or some tools, which could make some 
calculations for me ?

Thanks, 

B

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Re: [Asterisk-Users] so what are the additional hardware componentsneeded?

[Zoa]

You will need 1 tdm card in the server, with 1 or more fxo ports on it.

Thats all you will need. All pc will dial out through this 1 server.
Zoa.


Steve Totaro wrote:


keep reading

- Original Message -
*From:* infra struct mailto:[EMAIL PROTECTED]
*To:* asterisk-users@lists.digium.com
mailto:asterisk-users@lists.digium.com
*Sent:* Tuesday, June 07, 2005 10:02 PM
*Subject:* [Asterisk-Users] so what are the additional hardware
componentsneeded?

I have 20 personal computers in LAN with full duplex soundcard,
microphone(headset)
I will use this setup for making PC to PC phone calls

in addition I have a Linux server, in which i will be installing
asterisk

and

Internet Connection - DSL of speed 128kbps

I will be using asterisk for making calls to PSTN numbers(PC
to Phone calls)

so what are the additional hardware components needed?

i figured out,
Digium X100P (Asterisk, Linux server, X100P is for PSTN
connectivity (1 line) )
TDM400P on clients, in all Personal Computers (which also have
Softphones like SJPhone,As the softphones can also make PSTN calls)

Please users any comments about my finding?

Discover Yahoo!
Get on-the-go sports scores, stock quotes, news  more. Check it
out!
http://us.rd.yahoo.com/evt=32661/*http://discover.yahoo.com/mobile.html


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RE : [Asterisk-Users] Newbie on asterisk ask for configuratio help

[f6hqz-m]

Hi Roywish,

The best way is to publish here your .conf files to correct.

Good luck...

Best Regards,
Francois BERGERET,
Happy * french user  :-)

-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de craz sead
Envoyé : mercredi 8 juin 2005 09:45
À : asterisk-users@lists.digium.com
Objet : [Asterisk-Users] Newbie on asterisk ask for configuratio help


Hi all,

iam a student trying to build an asterisk pbx as a
simple configuration only two extention (using
Xlite)without outsite telephone line. i already follow
the instruction and seem the asterisk work fine
because there is no error message. when i configure
SIP.conf and extention.conf i hope the phone will ring
each other as an extention. but it doesn't work. i
follow the instruction at
www.automated.it/guidetoasterisk.htm this site but
nothing happen.

please help if someone have a configuration file or
any book that i can download and read because i am
really newbie here

thks

roywish 

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RE: [Asterisk-Users] SPA-2002 and NAT

[Chris Mason (Lists)]

Yes, I hooked one up yesterday. Although we have an Asterisk server in
house, I wanted to connected directly to a host in the US for Faxing. There
was no issue with NAT, and I did not do anything special beyond the usual.

[111]
callerid=test 111
type=friend
username=111
password=mine
host=dynamic
canreinvite=no
mailbox=111
dtmfmode=rfc2833
disallow=all
allow=ulaw
nat=yes
qualify=yes
context=default

With everything set to G711, faxing works...shocked me.


Chris Mason
www.anguillaguide.com
Tel:  (305) 704-7249 Fax: (815)301-9759  

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Waldo Rubinstein
 Sent: Tuesday, June 07, 2005 2:43 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [Asterisk-Users] SPA-2002 and NAT
 
 Does anyone have any experience with SPA-2002 behind a NAT 
 and working with Asterisk?
 
 Thanks,
 Waldo
 
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Re: [Asterisk-Users] DID on SIP channel

[Olle E. Johansson]

Joshua Colp wrote:
 You're actually confusing me when you say this due to the fact you're not
 giving much information, probably why nobody has responded yet. If the SIP
 server on the Nortel does an INVITE for the phone number, then asterisk will
 act accordingly and go to the phone number in the context you set for it.
 Note that if the Nortel is incapable of handling a challenge for
 credentials, you'll have to use a peer entry with insecure=very to match
 based on it's host/IP address.
 
 - Joshua Colp.
 (file in #asterisk on Freenode)
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of
 [EMAIL PROTECTED]
 Sent: Tuesday, June 07, 2005 7:12 PM
 To: asterisk-users@lists.digium.com
 Subject: [Asterisk-Users] DID on SIP channel
 
 Hi all.
 
 I need to implement the DID funcionality in a SIP channel with an ITSP. Is
 this possible to get it working!?
 
 The ITSP that im using has the alias feature in its SIP server(Nortel
 MCS5200), they provide just one register user/password and below this user
 they put a lot of other phone numbers.
 
 Ex.:
 register = 3030
 alias = 30302223
 alias = 30302224
 etc...
 
 Any clue for it!?
 
I guess you are registering with the Nortel SIP server? All the incoming
calls will go to the incoming extension you are registering with them.
If they add aliases for several incoming lines to one registration, you
need to check the To: header. This is only possible in CVS head with the
SIP get header function in the dial plan.

This is one of the reasons I am planning to implement a type=service
object in sip.conf

/O
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[Asterisk-Users] How to handle one incoming call on multiple lines?

[Erwin Lubbers]

Hi,

I have connected 4 analog public telephone lines to an Asterisk server using a 
Digium TDM400P card and that working fine. But my 4 lines are connected to 
each other in a group by the telecom operator. So if someone calls me all 4 
lines are ringing. I wrote a AGI script which will handle the incoming calls, 
but before it decides to answer the call or not the next channel is ringing 
and the script is started again. How can I create a situation that after the 
first ringing channel is coupled to a script the other channels are still 
ringing for the same call?

Regards,
Erwin
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Re: [Asterisk-Users] zaphfc: empty HDLC frame or bad CRC received

[Marco Parmeggiani]

Nick Barnes ha scritto:


I've only ever seen when the signalling is wrong. For example if the line is
in PTMP mode when it should be in PTP or vice-versa.



this is the zapata.conf:
group = 1
context=default
signalling = bri_net_ptmp
channel = 1-2



So, you're using NT mode PTMP signalling.

Is the Asterisk box plugging into an ISDN circuit provided by a telco? If it
is, then use bri_cpe_ptmp (for Point to MultiPoint) or bri_cpe (for
Point to Point) instead of bri_net_ptmp. If it's plugged into a different
ISDN device and needs to be in NT mode, then try bri_net instead.



you pointed me in the right direction.
the card is connected directly to the telco isdn and it should run in TE 
mode, also the signalling should be bri_cpe_ptmp


Thanks
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[Asterisk-Users] Faxing error rtp.c:504 ast_rtp_read: Unknown RTP codec 100 received

[Chris Mason (Lists)]

When I am receiving faxes, which will go through a Sipura 2002, the server
says 

rtp.c:504 ast_rtp_read: Unknown RTP codec 100 received

I still get the fax, any idea what this is?

Chris Mason


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Re: [Asterisk-Users] English vs American voice files

[Andrew Thrift]

I also have someone in New Zealand who has done some for our own
Asterisk server.

Mark Phillips wrote:

 I've found a woman whom is happy to help make English voice files!
 Ironic that she should be in New Zealand.

 More when I have the files.


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Re: [Asterisk-Users] DID on SIP channel

[Olle E. Johansson]

Joshua Colp wrote:
 Okay lemme give you something that should work some magic!
 
 Stuff for sip.conf:
 [nortel]
 type=peer
 host=IP ADDRESS OF NORTEL
 disallow=all
 allow=ulaw 
 context=inbound_nortel
 insecure=very
 
 Stuff for extensions.conf:
 [inbound_nortel]
 exten = 3030,1,Dial(SIP/whatever)
 exten = 30302223,1,Dial(SIP/bleh)
 ... SO ON...
 
 Use your head to figure out some of the stuff for what you should put in.

This is a great configuration if you do not register. I would propably
add acl controls to avoid matching anyone using the name nortel in
communication.

/O
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Re: [Asterisk-Users] AT-320 + supervised transfer

[Gavin Hamill]

On Tuesday 07 June 2005 09:44, Giordano Grandis wrote:
 Ok, just a thing...cuold is see a sample peer in tuou extensions.conf

I'm newly testing the atxfer and i always the same question: if i
 transfer a call to a peer that don't answer me, ho can i re-take the
 call. Actually i got the call hanged up without the possibility the
 speack back with my first caller.

I have the same problem now that I've actually tried this... and so have other 
people - check this thread which has been running at the same time:

http://lists.digium.com/pipermail/asterisk-users/2005-June/110856.html

The 'hook flash' certainly doesn't have any effect - it just puts the call on 
hold (even though it's already on hold because of the atxfer..)

sigh

Cheers,
Gavin

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Re: [Asterisk-Users] Asterisk to Avaya PBX using TDM cards

[Rich Adamson]

  I'm new in this field, have been reading a lot, and have a little question. 
 could it be possible to connect an Avaya IP office pbx to asterisk using a 
 E1/T1/Pri?
 
 Original instalation:
 
 Telefone company|Pri---Pri|IP Pffice
 
 My Question:
 
 Telefone company|Pri ---TDM|Asterisk|TDM ---Pri|IP Office
 
 I know that it can be done by using h323, but I need a card on the IPOffice 
 my 
 problem is that I have no more room for expantion on this pbx so I was 
 thinking instead of upgrading the IPOffice maybe I can start using *. 
 The secret of success is converting your problems into opportunities

Very doable as long as the existing pbx is working via the PRI. Lots
of options in terms of exactly how you configure asterisk.


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[Asterisk-Users] Station Lines

[Sean Cook]

I am not sure if this is really possible but I figured I would ask
anyway.  I have a customer who wants an asterisk system.  Currently they
have a BizFon system.

The feature that he really wants is to be able to pick up any line and
have all the stations show up on his phone.  Is this possible in
asterisk?  If so can someone point me in the right direction?

Sean

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Re: [Asterisk-Users] D-link DPH-80 (SIP) call to asterisk problem

[Eugene Crosser]

Followup to myself:

 I have a D-Link DPH-80S SIP phone (it's a non-US model), and I am trying
 to make it work with Asterisk.  I tried versions 1.0.7 and yesterday's
 CVS and the behavior is the same.
 
 The phone registers with no problem, and can accept calls.
 
 But when I try to make outgoing call, there is a series of invite
 requests from the phone, to which asterisk responds with 407.  Comparing
 logs with that of a soft phone, the difference is as follows:
 
 Softphone
 p: INVITE w/o Auth
 a: 407 auth required
 p: ACK
 p: INVITE with auth
 a: 200 OK
 - all of these share the same Call-ID.
 
 D-link:
 p: INVITE w/o Auth
 a: 407 auth required
 p: ACK
 p: INVITE with auth and new Call-ID
 a: 407 auth required
 p: ACK
 p: INVITE with auth and new Call-ID
 a: 407 auth required
 p: ACK
 p: INVITE with auth and new Call-ID
 - et cetera
 
 Apparently when Call-ID is new asterisk no longer matches the nonce that
 it sent to this phone (check_auth is called with NULL third argument).
 
 This does look like a bug in the phone firmware.  However, the phone can
 successfully initiate calls via several commercial and community
 providers.  I tried iconnecthere.com and voipuser.org and it works!
 
 Now, the question: could it be possible to make the phone work with
 asterisk?  Any ideas?  I can send the log on request (it is rather big).

OK, I figured a workaround.  I have to create two separate entries in
sip.conf instead of one entry with type=friend:

[555-in]
; D-Link hard phone
type=user
context=home
host=dynamic
insecure=very
canreinvite=no
callerid=D-link Phone555

[555]
; D-Link hard phone
type=peer
context=home
host=dynamic
user=555
secret=555dlink
canreinvite=no

type=peer entry allows registration of the phone (because DPH-80S refuse
to work if it is not registered), and type=user entry tells asterisk to
accept INVITE from the phone without authentication.  Now my phne
actually works for both incoming and outgoing!

Eugene


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[Asterisk-Users] newbie question

[Charles Austin]

Greetings,

I have my first asterisk installation up and running, thanks to a lot
of reading.  Could anyone point me in the direction of things to read
on automated outbound dialing?  NOT predictive dialing - I will not
have agents handling the calls.  These calls are reminders for
appointments, etc.

Thanks!
Charles
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Re: [Asterisk-Users] How to handle one incoming call on multiple lines?

[Julian J. M.]

Isn't it easier to talk to your Telco, and tell them to just ring the
first free line, instead of all 4?

Julian J. M.

On 6/8/05, Erwin Lubbers [EMAIL PROTECTED] wrote:
 Hi,
 
 I have connected 4 analog public telephone lines to an Asterisk server using a
 Digium TDM400P card and that working fine. But my 4 lines are connected to
 each other in a group by the telecom operator. So if someone calls me all 4
 lines are ringing. I wrote a AGI script which will handle the incoming calls,
 but before it decides to answer the call or not the next channel is ringing
 and the script is started again. How can I create a situation that after the
 first ringing channel is coupled to a script the other channels are still
 ringing for the same call?
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Re: [Asterisk-Users] Help Connecting Cisco AS5300 to Asterisk

[Marcelo Pacheco]

Stefan,

Is it possible to have the Cisco forward calls between T1 or E1 interefaces, 
without VOIP DSPs, but only Modem DSPs ?

I need to have an AS5350 that is currently configured as a dial-in RAS to 
forward incoming calls to Asterisk, but I can't do it with SIP, as I don't 
have VOIP DSPs on the AS5350, only modem DSPs.

TIA,

Marcelo Pacheco

Em Qua 08 Jun 2005 06:36, chawki hammoud escreveu:
 I have already looked into this page.  I thought this
 was for AS 5350, I am not familiar with Cisco products
 and I don't know if there is a difference. And there
 is no Asterisk set-up in this example.

 Regards;

 --- Stefan Reuter [EMAIL PROTECTED] wrote:
   It doesn't have to be IAX. Do you know how to
   configure it with another protocol?
 
  have a look at

 http://ertw.com/blog/archives/asterisk_and_an_as5350_sip_peer-190405.html

  =Stefan
 
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Re: [Asterisk-Users] Station Lines

[Walt Reed]

On Wed, Jun 08, 2005 at 08:38:27AM -0400, Sean Cook said:
 The feature that he really wants is to be able to pick up any line and
 have all the stations show up on his phone.  Is this possible in
 asterisk?  If so can someone point me in the right direction?

That describes a key system. Asterisk is a PBX. Trying to make Asterisk
function like a key system (while possible) is difficult, and will
result in much frustration. 

Instead, you are best off showing this person how to use a PBX properly.
It may take a little getting used to, but it's best in the long run
because you won't be supporting a goofy system.


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Re: [Asterisk-Users] IAXtel update!

[Kevin P. Fleming]

Rich Adamson wrote:


Any chance that we could get someone to implement the milliwatt
generator and echo test number. Would be kind of handy for testing
various items (eg, jitterbuffer).


It's running CVS HEAD (which means it has the new jb since we didn't 
disable it, but then again it's all VOIP so the jb doesn't get enabled 
anyway), with Realtime for IAX2 friends and the experimental hashtable 
config parsing code. If you can email me or Russell with what you think 
should be enabled there we'll see what we can do.

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RE: [Asterisk-Users] Books

[The VoIP Connection]

We have it:

http://www.thevoipconnection.com/store/catalog/product_16198_VoIP_Telephony_
with_Asterisk_by_Paul_Mahler.html

Michael Crown
Managing Partner
The VoIP Connection
321.989.6728 ext. 611
sip:[EMAIL PROTECTED]
 

 -Original Message-
 From: John H [mailto:[EMAIL PROTECTED] 
 Sent: Wednesday, June 08, 2005 12:05 AM
 To: asterisk-users@lists.digium.com
 Subject: [Asterisk-Users] Books
 
 Hello all, I was wondering if anyone know where i can find a 
 book on Asterisk, i have been told about VoIP With Asterisk 
 but i  am unsure where to find it, any ideas plase?
 
 

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Re: [Asterisk-Users] Station Lines

[Andrew Kohlsmith]

On Wednesday 08 June 2005 08:38, Sean Cook wrote:
 I am not sure if this is really possible but I figured I would ask
 anyway.  I have a customer who wants an asterisk system.  Currently they
 have a BizFon system.

 The feature that he really wants is to be able to pick up any line and
 have all the stations show up on his phone.  Is this possible in
 asterisk?  If so can someone point me in the right direction?

Pick up any line: yes.  All stations showing up?  If your phone has multiple 
line appearances and you hint them properly, yes.  Alternatively you could 
use the asterisk management portal software but that runs on a PC.

I worked with the old BizFon systems...   Nasty nasty nasty.

-A.
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Re: [Asterisk-Users] English vs American voice files

[Paul Redstone]

Hi

In the end we found it easy to record our own using this section in 
extensions.conf. This also meant that we could add our own company specific 
ones in the same voice (not shown here). Basically you get someone to dial the 
8NNN1 to record or 8NNN2 to playback. The prompts are shown below and we just 
printed out this text. It was our intention to use festival to read these, but 
this was easier. The text has been amended to reflect the UK (e.g. Hash instead 
of pound).

Many sites may not need all of them and if you omit them the US voice will play 
instead. 

Paul
 [EMAIL PROTECTED]



[macro-record-message]
;
; ARG1 file name of message, assumed to be in sounds folder, but if below has a 
subfolder name prepended
; ARG2 text describing message
; Called with 8NNNX where NNN is the message and X is 1 to playback or 2 to 
record.
exten = s,1,GotoIf($[${MACRO_EXTEN:4} = 2]?10:2) ; if fifth digit is 2 
then go to record, otherwise playback
exten = s,2,Playback(/var/lib/asterisk/sounds/${ARG1}) ;playback here
exten = s,3,Wait(1)
exten = s,4,Hangup
exten = s,10,Wait(1) ;record here
exten = s,11,Record(/var/lib/asterisk/sounds/${ARG1}:gsm)
exten = s,12,Wait(1)
exten = s,13,Playback(/var/lib/asterisk/sounds/${ARG1})
exten = s,14,Wait(1)
exten = s,15,Hangup

[record-messages]
; Special context used to record voicemail messages
exten = _8001X,1,Macro(record-message,gb/hours,hours)
exten = _8002X,1,Macro(record-message,gb/minutes, minutes) 
exten = _8003X,1,Macro(record-message,gb/auth-incorrect, Password incorrect. 
Please enter your password followed by the hash key)
exten = _8004X,1,Macro(record-message,gb/auth-thankyou, Thank you. )
exten = _8005X,1,Macro(record-message,gb/invalid, 'I am sorry, that is not a 
valid extension. Please try again' )
exten = _8006X,1,Macro(record-message,gb/pbx-invalid, 'I am sorry, that's not 
a valid extension. Please try again. ')
exten = _8007X,1,Macro(record-message,gb/pbx-invalidpark, 'I am sorry, there 
is no call parked on that extension. Please try again.') 
exten = _8008X,1,Macro(record-message,gb/pbx-transfer, Transfer. )
exten = _8009X,1,Macro(record-message,gb/privacy-incorrect, 'I'm sorry, that 
number is not valid. ')
exten = _8010X,1,Macro(record-message,gb/privacy-prompt, (Please enter your 
ten-digit phone number, starting with the area code. )
exten = _8011X,1,Macro(record-message,gb/privacy-thankyou, Thank you. )
exten = _8012X,1,Macro(record-message,gb/privacy-unident, The party you are 
trying to reach does not accept unidentified calls. )
exten = _8013X,1,Macro(record-message,gb/ss-noservice, The number you have 
dialed is not in service. Please check the number and try again. )
exten = _8014X,1,Macro(record-message,gb/transfer, Please hold while I try 
that extension. )
exten = _8015X,1,Macro(record-message,gb/tt-allbusy, All representatives of 
the household are currently assisting other telemarketers. Please hold and your 
call will be answered in the order it was received. )
exten = _8016X,1,Macro(record-message,gb/tt-monkeysintro, They have been 
carried away by monkeys. )
exten = _8017X,1,Macro(record-message,gb/tt-somethingwrong, Something is 
terribly wrong, )
exten = _8018X,1,Macro(record-message,gb/tt-weasels, Weasels have eaten our 
phone system. )
exten = 80191,1,Playback(/var/lib/asterisk/sounds/gb/tt-allbusy)
exten = 80191,2,Playback(/var/lib/asterisk/sounds/gb/tt-monkeysintro)
exten = 80191,3,Playback(/var/lib/asterisk/sounds/tt-monkeys)  ; Ho Ho 
 
exten = _8020X,1,Macro(record-message,gb/dir-instr, 'If this is the person you 
are looking for press 1 now, otherwise please press star now. )
exten = _8021X,1,Macro(record-message,gb/dir-intro, 'Welcome to the directory. 
Please enter the first three letters of your party's last name using your touch 
tone keypad. Use the 7 key for Q, and the 9 key for Zed.') 
exten = _8022X,1,Macro(record-message,gb/dir-nomatch, No directory entries 
match your search. )
exten = _8023X,1,Macro(record-message,gb/dir-nomore, There are no more 
compatible entries in the directory. )
;; Not needed - blank exten = _8024X,1,Macro(record-message,gb/dir-intro-fn, 
TO BE FILLED IN)
exten = _8031X,1,Macro(record-message,gb/conf-getchannel, Please enter the 
channel number followed by the hash key. )
exten = _8032X,1,Macro(record-message,gb/conf-getconfno, Please enter your 
conference number followed by the hash key. )
exten = _8033X,1,Macro(record-message,gb/conf-getpin, Please enter the 
conference pin number. )
exten = _8034X,1,Macro(record-message,gb/conf-invalid, That is not a valid 
conference number. Please try again. )
exten = _8035X,1,Macro(record-message,gb/conf-invalidpin, That pin is invalid 
for this conference. )
exten = _8036X,1,Macro(record-message,gb/conf-onlyperson, You are currently 
the only person in this conference. )
exten = _8037X,1,Macro(record-message,gb/conf-adminmenu, 'Please press 1 to 
mute or unmute yourself. Or press 2 to lock or unlock the conference. )
exten = 

Re: [Asterisk-Users] SS7

[Kevin P. Fleming]

Matt wrote:


Isn't the SS7 code for Asterisk available under the commercial
Asterisk license and that's the only way to get it?


No, that's a poor description of the availability... one of these days 
I'll have to ask them to stop wording it in quite that way.


If you want to use the commercial SS7 stack that exists for Asterisk, 
you can _only_ use it with a commercially licensed copy of Asterisk 
provided by that same vendor. The SS7 stack is _not_ a Digium product 
nor do we have any involvement with it, other than allowing the SS7 
stack vendor to provide commercially licensed copies of Asterisk for use 
with it.

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Re: [Asterisk-Users] English vs American voice files

[Mark Phillips]

I think you miss the point Andrew. She's not from NZ but from England. 
She speaks English. Says six and not sex etc.


Mark

Andrew Thrift wrote:


I also have someone in New Zealand who has done some for our own
Asterisk server.

Mark Phillips wrote:

 


I've found a woman whom is happy to help make English voice files!
Ironic that she should be in New Zealand.

More when I have the files.

   



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--

Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com

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RE: [Asterisk-Users] error message: INIT: Id s0 respawning toofast:disable for 5 minutes

[Justin Ellison]

I had this same issue - it's because AAH tries to run a getty on ttyS0,
and if you have COM1 disabled in the bios (or it doesn't exist), this
won't work.  If you're getting this issue, edit /etc/inittab, and
comment out the line that says:

s0:12345:respawn:/sbin/agetty -i -h -L 9600 ttyS0 vt100

Then save the file, and do a 'killall -HUP init' as root - the problem
should go away.

Justin

On Wed, 2005-06-08 at 08:54 +0100, Wagner Gimenes wrote:
 Guys (and Gals),
 
 FYI I also have the *same* message here. Wonder is it is related to my
 Compaq D500 Space Saver PIV 1.7 or the fact that I don't yet have a
 modem card in the * box.
 
 (Please don't shoot me, did try Google first)
 
 Many thanks,
 
 Wagner Gimenes 

-- 

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Re: [Asterisk-Users] CallerID/chan_sccp

[Sergio Chersovani]

Joseph ha scritto:

When sending a call to a line defined on chan_sccp, there is an error 
on the console that says:


Jun  7 08:22:29 WARNING[3924]: sccp_channel.c:79
sccp_channel_send_callinfo: Incoming call SCCP/Line1-0008 doesn't
have CallerId name
   


Fixed, you can find the patch here
http://www.c-net.it/chan_sccp/

Sergio Chersovani
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Re: [Asterisk-Users] English vs American voice files

[Sahil Gupta]

Like to share who can record NZ / Australian voices?

Regards,


Sahil Gupta
VoiceValley

On Wed, 8 Jun 2005, Mark Phillips wrote:

I think you miss the point Andrew. She's not from NZ but from England. She 
speaks English. Says six and not sex etc.


Mark

Andrew Thrift wrote:


I also have someone in New Zealand who has done some for our own
Asterisk server.

Mark Phillips wrote:



I've found a woman whom is happy to help make English voice files!
Ironic that she should be in New Zealand.

More when I have the files.




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--

Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com

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[Asterisk-Users] Fax + Fritz + Capi + detection

[sylvain garcia]

Hello

I'm newbie in asterisk and i have a AVM Audiovisuelles MKTG  Computer System 
GmbH Fritz!PCI v2.0 ISDN (rev 02) with CAPI Driver.
I would like install fax detection, but i don't know if i should use  
NVBackground detect; or CapiAnswerFAx; or other.
I don't understantd operation of fax.

Tx

ps: sorry for English i'm french




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Re: [Asterisk-Users] Books

[Zoa]

I suggest you wait a little for the new o'reilly book about asterisk.
Amazon already accepts pre-orders for it

The VoIP Connection wrote:


We have it:

http://www.thevoipconnection.com/store/catalog/product_16198_VoIP_Telephony_
with_Asterisk_by_Paul_Mahler.html

Michael Crown
Managing Partner
The VoIP Connection
321.989.6728 ext. 611
sip:[EMAIL PROTECTED]





-Original Message-
From: John H [mailto:[EMAIL PROTECTED]
Sent: Wednesday, June 08, 2005 12:05 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Books

Hello all, I was wondering if anyone know where i can find a
book on Asterisk, i have been told about VoIP With Asterisk
but i  am unsure where to find it, any ideas plase?






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[Asterisk-Users] Polycom 500 Group Call Pickup Feature and *

[Chris Coulthurst]

If you activate (via sip.cfg) the feature Group Call Pickup, its no
surprise that asterisk doesn't know what to do with this feature
request.  But it is being sent as a SIP SUBSCRIBE request, and I'm
wondering if, as asterisk stands, there is a way to take advantage of
this feature to emulate the *8# normal behavior.


If anyone has any input, there is also a call parking function that I
think is SIP SUBSCRIBE-based.


Here is the 'sip debug' snippet from when I pressed the New Call -
Pickup - Group softkeys:


Sip read: 
SUBSCRIBE sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.234;branch=z9hG4bKa58a6cc24AEA0129
From: Chris Office sip:[EMAIL PROTECTED];tag=569A308-31C12E4D
To: sip:[EMAIL PROTECTED]
CSeq: 1 SUBSCRIBE
Call-ID: [EMAIL PROTECTED]
Contact: sip:[EMAIL PROTECTED]
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE,
NOTIFY, PRACK, UPDATE, REFER
Event: dialog
User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.5.2.0054
Accept: application/dialog-info+xml
Max-Forwards: 70
Expires: 0
Content-Length: 0


14 headers, 0 lines
Using latest SUBSCRIBE request as basis request
Sending to 192.168.0.234 : 5060 (non-NAT)
Found peer '201'
Transmitting (no NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.0.234;branch=z9hG4bKa58a6cc24AEA0129
From: Chris Office sip:[EMAIL PROTECTED];tag=569A308-31C12E4D
To: sip:[EMAIL PROTECTED];tag=as1b873db6
Call-ID: [EMAIL PROTECTED]
CSeq: 1 SUBSCRIBE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:[EMAIL PROTECTED]
Proxy-Authenticate: Digest realm=asterisk, nonce=5041eff0
Content-Length: 0


 to 192.168.0.234:5060
Scheduling destruction of call '[EMAIL PROTECTED]'
in 15000 ms
morse*CLI 

Sip read: 
SUBSCRIBE sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.234;branch=z9hG4bK802f53579213D6EA
From: Chris Office sip:[EMAIL PROTECTED];tag=569A308-31C12E4D
To: sip:[EMAIL PROTECTED]
CSeq: 2 SUBSCRIBE
Call-ID: [EMAIL PROTECTED]
Contact: sip:[EMAIL PROTECTED]
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE,
NOTIFY, PRACK, UPDATE, REFER
Event: dialog
User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.5.2.0054
Accept: application/dialog-info+xml
Proxy-Authorization: Digest username=201, realm=asterisk,
nonce=5041eff0, uri=sip:[EMAIL PROTECTED]:5060,
response=b48b989d85958a6ce18c9431058ce6f3, algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


15 headers, 0 lines
Using latest SUBSCRIBE request as basis request
Sending to 192.168.0.234 : 5060 (non-NAT)
Found peer '201'
Looking for groupcallpickup in default
Transmitting (no NAT):
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.0.234;branch=z9hG4bK802f53579213D6EA
From: Chris Office sip:[EMAIL PROTECTED];tag=569A308-31C12E4D
To: sip:[EMAIL PROTECTED];tag=as1b873db6
Call-ID: [EMAIL PROTECTED]
CSeq: 2 SUBSCRIBE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:[EMAIL PROTECTED]
Content-Length: 0


 to 192.168.0.234:5060
Destroying call '[EMAIL PROTECTED]'
morse*CLI sip no debug
SIP Debugging Disabled

Chris Coulthurst
[EMAIL PROTECTED]
 


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[Asterisk-Users] sip to sip echo with meetme, timing

[Jerry Bonner]

When calling from sip phone to sip phone ( cisco 7940 ) we have very little or 
no echo. When conferencing through meetme through a sip only server, we 
experience lots of echo.


Would this have anything to do with the timing 
source?


The server is using ztdummy on 2.4 with uhci usb. Would using 
digium hardware timing help with this? Or switching to 2.6?


first time post, thanks for your comments / suggestions...

~jerry
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[Asterisk-Users] * @ Home: All Circuits busy

[maoleson]

All,

I have an [EMAIL PROTECTED] installation with a TDM40B card.  I can make 
internal 
IP calls with no problems, but when I try to dial out I get a message that “All 
Circuits are Busy”.  I looked into the Zapata.conf files and such but see no 
modifications.  Is there a step that I am missing??  Does anyone have 
documentation of step-by-step config for this TDM40B card?

Thanks,
Marc 
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Re: [Asterisk-Users] How to handle one incoming call on multiple lines?

[Erwin Lubbers]

Julian,

Thanks, but it isn't an option because the Telco is actually connected to
a PBX which is connected to Asterisk which should act as a intelligent
answering device during non-office hours. The PBX isn't capable of doing
this. Any other option?

Regards,
Erwin

 Isn't it easier to talk to your Telco, and tell them to just ring the
 first free line, instead of all 4?

 Julian J. M.



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Re: [Asterisk-Users] How to handle one incoming call on multiple lines?

[Andrew Latham]

unplug the other three lines 

This is an after hours ring group or is this enabled after hours only?

On 6/8/05, Erwin Lubbers [EMAIL PROTECTED] wrote:
 Julian,
 
 Thanks, but it isn't an option because the Telco is actually connected to
 a PBX which is connected to Asterisk which should act as a intelligent
 answering device during non-office hours. The PBX isn't capable of doing
 this. Any other option?
 
 Regards,
 Erwin
 
  Isn't it easier to talk to your Telco, and tell them to just ring the
  first free line, instead of all 4?
 
  Julian J. M.
 
 
 
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[Asterisk-Users] TDM400P... ignoring hanguponpolarityswitch

[Neil and Fiona]

I've just had polarity reversal provisioned by our telco to test hangup
detect with a TDM400P

I've set hanguponpolarityswitch=yes in zapata.conf

When I start Asterisk I get ignoring hanguponpolarityswitch
in /var/log/asterisk/messages

I assume that the option is either not valid or conflicts with another
setting somewhere.

Any ideas?

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RE: [Asterisk-Users] How to handle one incoming call on multiple lines?

[Florian Overkamp]

Hi, 

 -Original Message-
 Thanks, but it isn't an option because the Telco is actually 
 connected to
 a PBX which is connected to Asterisk which should act as a intelligent
 answering device during non-office hours. The PBX isn't 
 capable of doing
 this. Any other option?

Hmm, this is a bit of a hack, but it might suit your needs:

- Make sure each of those lines goes into a different extension or context
- Add a delay on each line, like this:

exten = line1,1,Do stuff

exten = line2,1,Wait(2)
exten = line2,1,Do stuff

exten = line3,1,Wait(4)
exten = line3,1,Do stuff

exten = line4,1,Wait(6)
exten = line4,1,Do stuff

Could this help your case ?

Florian


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[Asterisk-Users] Latest CVS and app_rxfax

[Dave Cotton]

With the current CVS-HEAD line 88 of app_rxfax.c causes an error.

#if (ASTERISK_VERSION_NUM = 010300)
  chan-callerid,

app_rxfax.c:88: error: 'struct ast_channel' has no member named
'callerid'

Commenting out the if else combination of course gives a clean compile.




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Re: [Asterisk-Users] * @ Home: All Circuits busy

[Dean Mumby]

[EMAIL PROTECTED] wrote:


All,

I have an [EMAIL PROTECTED] installation with a TDM40B card.  I can make internal 
IP calls with no problems, but when I try to dial out I get a message that All 
Circuits are Busy.  I looked into the Zapata.conf files and such but see no 
modifications.  Is there a step that I am missing??  Does anyone have 
documentation of step-by-step config for this TDM40B card?


Thanks,
Marc 
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have you run the genzaptelconf -s
try aah-help for more info


Dean



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Re: [Asterisk-Users] Help Connecting Cisco AS5300 to Asterisk

[chawki hammoud]

--- Alexander Ilyushin [EMAIL PROTECTED] wrote:

 You can connect it only via SIP
 
If you know how to configure the cisco AS5300 and SIP,
I appreciate it if you write the configuration down.

Thanks;



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Re: [Asterisk-Users] newbie question

[Moises Silva]

read in voip-info.org about Asterisk Call Manager API, and may be an
easier soultion are the .call files that you can pleace in
/var/spool/asterisk/outgoing/ these files have a description of the
type of call you wanna make, in the very moment that you place the
file there, a call will be Originated automagically. With Asterisk
Call Manager API you can do something similar ( or equal ) using the
Originate action.

best regards

On 6/8/05, Charles Austin [EMAIL PROTECTED] wrote:
 Greetings,
 
 I have my first asterisk installation up and running, thanks to a lot
 of reading.  Could anyone point me in the direction of things to read
 on automated outbound dialing?  NOT predictive dialing - I will not
 have agents handling the calls.  These calls are reminders for
 appointments, etc.
 
 Thanks!
 Charles
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[Asterisk-Users] Clicks in audio with TE100P PRI

[Alejandro G]

Hi, I have a problem I will describe. I have PAP2 connected to the internet
to an asterisk box with 2 TDM cards, one TE100P E1 with PRI and one TDM400P
with 2 FXS an one FXO.

When I call to the TDM400 cards from the PAP2 eveything is OK, sound quality
is perfect.
When I call to terminate the call in PSTN through E100P I hear clicks which
aparently are RTP packet looses. This clicks are only heard in the PSTN
side, not in the PAP2.

If I connect PAP2 in LAN to the *, everything sounds is normal. So I
evaluate the following:

1. Delay or something similar in internet could not be the problem because
it works with TDM400P (same configuration)
2. The PAP2 could not be the problem because it works with TDM400 (and other
ip phones) and in a LAN.
3. The TE100P could not be the problem because it works fine if the PAP2 is
connected via lan and not via internet.
4. With other IP phones everything works fine.

It seems that the combination of PAP2 - Internet - TE100P is the
problem. Any suggestions?
is there any jitter buffer adjust for the sip channel or zap in the * side
only for the TE100P? I look that in zapata.conf there is a jitterbuffer
parameters which defaults to 4, should I modify it?

Thanks,


Alejandro G.








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Re: [Asterisk-Users] How to handle one incoming call on multiple lines?

[Henry Coleman]

This feature is called  attendant - night answer position. Is it not 
possible to switch the incoming call to an alternate extension based on 
time of day ?


Henry

Florian Overkamp wrote:

Hi, 

 


-Original Message-
Thanks, but it isn't an option because the Telco is actually 
connected to

a PBX which is connected to Asterisk which should act as a intelligent
answering device during non-office hours. The PBX isn't 
capable of doing

this. Any other option?
   



Hmm, this is a bit of a hack, but it might suit your needs:

- Make sure each of those lines goes into a different extension or context
- Add a delay on each line, like this:

exten = line1,1,Do stuff

exten = line2,1,Wait(2)
exten = line2,1,Do stuff

exten = line3,1,Wait(4)
exten = line3,1,Do stuff

exten = line4,1,Wait(6)
exten = line4,1,Do stuff

Could this help your case ?

Florian


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Re: [Asterisk-Users] no DTMF pass-thru

[Moises Silva]

make sure that the DTMF mode configuration in Asterisk match the
configuration inside the Grandstream devices. I mean, in asterisk
config you may need something like

[20]
type=friend
.blah
dtmfmode=info

and of inside the configuration of the Grandstream device you may have
to use the same dtmfmode.

give that a try, hope it helps you.

best regards

On 6/8/05, Asterisk [EMAIL PROTECTED] wrote:
 Hi all,
 
 We have a little problem.
 One of our customers has a problem with DTMF pass-thru.
 They use GrandStream 286 devices to connect their pstn phones to asterisk.
 everything works like a charm, except DTMF pass-thru. when they call an IVR
 system, they cannot select options because the DTMF tones never reach the
 IVR.
 Normal asterisk voicemail works though. so it looks like asterisk is not
 forwarding the DTMF tones to the zap interface.
 Is this a setup problem or asterisk intended?
 
 Regards.
 Andre
 
  
  
 -
 Disclaimer
 
 This email and any files transmitted with it are confidential and intended
 solely for the use of the individual or entity to whom they are addressed.
 If you have received this email in error please notify the system manager.
 Please note that any views or opinions presented in this email are solely
 those of the author and do not necessarily represent those of the company.
 Finally, the recipient should check this email and any attachments for the
 presence of viruses. The company accepts no liability for any damage caused
 by any virus transmitted by this email.
 
  
  
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Re: [Asterisk-Users] TDM400P... ignoring hanguponpolarityswitch

[Andrew Kohlsmith]

On Wednesday 08 June 2005 10:57, Neil and Fiona wrote:
 I've set hanguponpolarityswitch=yes in zapata.conf

Do you also have the signaling on the channel set to kewlstart?  I don't 
believe polarity detection does anything without this signaling type.

 When I start Asterisk I get ignoring hanguponpolarityswitch
 in /var/log/asterisk/messages

When you start, or when you reload asterisk?

 Any ideas?

Well, you can start by telling us the version of asterisk you're running, and 
the date of the CVS pull if it's from CVS.  You could also post your 
zapata.conf and zaptel.conf files. 

You really didn't leave us a lot to go on.

-A.
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Re: [Asterisk-Users] rxfax not answering

[JD Austin]

rxfax doesnt work with voip, you need something like NVFaxDetect from 
Newman Telecom to detect the incoming fax.
Essentially you sent him an email and he'll send you the code.  Once you 
compile them into asterisk you can add it.

http://www.voip-info.org/tiki-index.php?page=NVFaxDetect
JD
Antonio Gallo wrote:


Hello i would like to receive incoming faxes thru' asterisk as tiff
files thru' the rxfax application.

I setup extensions 101 like this
exten= 101,1,rxfax(/tmp/fax.tif)

then from CLI i run:
dial 101
and rxfax send me his scream about the fax ^^

instead when i send a real fax from a faxmachine to that extension
my 101+rxfax is executed but it just does nothing

the call is originated by a FAX on PSTN and received via VoIP by
asterisk using a/u law codec

i think that is my VoIP provider that has some fax problem.

Is this the problem or there maybe other solutions?

Thank you, Antonio


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RE: [Asterisk-Users] * @ Home: All Circuits busy

[maoleson]

Dean,

Actually, I have run genzaptelconf -s -d but it still didn’t seem to modify any 
of the config files that I look at in the AMP console.  Should I try modifying 
the config files manually?
Thanks,
Marc
-Original Message-
From:   [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED]  On Behalf Of Dean Mumby
Sent:   Wednesday, June 08, 2005 10:07 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:Re: [Asterisk-Users] * @ Home: All Circuits busy

[EMAIL PROTECTED] wrote:
All,

I have an [EMAIL PROTECTED] installation with a TDM40B card.  I can make 
internal 
IP calls with no problems, but when I try to dial out I get a message that “All 
Circuits are Busy”.  I looked into the Zapata.conf files and such but see no 
modifications.  Is there a step that I am missing??  Does anyone have 
documentation of step-by-step config for this TDM40B card?

Thanks,
Marc 
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have you run the genzaptelconf -s
try aah-help for more info


Dean


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Re: [Asterisk-Users] Clicks in audio with TE100P PRI

[Andrew Kohlsmith]

On Wednesday 08 June 2005 11:19, Alejandro G wrote:
 When I call to the TDM400 cards from the PAP2 eveything is OK, sound
 quality is perfect.
 When I call to terminate the call in PSTN through E100P I hear clicks which
 aparently are RTP packet looses. This clicks are only heard in the PSTN
 side, not in the PAP2.

You just described a classic clock slip situation.

Are you synchronizing to the PSTN?  i.e. does your span line have '1' for 
clocking?  You want to sync to them instead of free-run (clock of '0').

-A.
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Re: [Asterisk-Users] How to handle one incoming call on multiple lines?

[Andrew Kohlsmith]

On Wednesday 08 June 2005 11:24, Henry Coleman wrote:
 This feature is called  attendant - night answer position. Is it not
 possible to switch the incoming call to an alternate extension based on
 time of day ?

You need to read up.  This exact situation is given in the Asterisk Handbook.

http://www.digium.com/handbook-draft.pdf

In particular, you want GotoIfTime().

-A.
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Re: [Asterisk-Users] How to handle one incoming call on multiple lines?

[Henry Coleman]

Yeh, this is called line hunting  all telco's offer this... you get  
one published number but say 12 lines each line actually has a 
number but just calling the main number will automatically roll-over to 
the first available line in that hunting group. By the way, outgoing 
calls that use the same lines should have hunting groups going in the 
opposite direction (for obvious reasons).
Unfortunatly, for those who want to develop ACD (Automatic Call 
Distribution) 
this mode is useless, if you were to distribute calls based on this 
method the person attached to the first line would get most of the calls 
while the last would be able to put their feet up and whistle dixie


Have fun ..Henry
 
 


Erwin Lubbers wrote:


Julian,

Thanks, but it isn't an option because the Telco is actually connected to
a PBX which is connected to Asterisk which should act as a intelligent
answering device during non-office hours. The PBX isn't capable of doing
this. Any other option?

Regards,
Erwin

 


Isn't it easier to talk to your Telco, and tell them to just ring the
first free line, instead of all 4?

Julian J. M.

   




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Re: [Asterisk-Users] TDM400P... ignoring hanguponpolarityswitch

[Neil and Fiona]

On Wed, 2005-06-08 at 11:34 -0400, Andrew Kohlsmith wrote:
 On Wednesday 08 June 2005 10:57, Neil and Fiona wrote:
  I've set hanguponpolarityswitch=yes in zapata.conf
 
 Do you also have the signaling on the channel set to kewlstart?  I don't 
 believe polarity detection does anything without this signaling type.
Yes. singnalling=fxs_ks

4 fxo modlues load in driver successfully.


 
  When I start Asterisk I get ignoring hanguponpolarityswitch
  in /var/log/asterisk/messages
 
 When you start, or when you reload asterisk?
When starting asterisk

/var/log/messages seems to be indicating that the wctdm driver thinks
that the polarity of the line is reversed on start. (ie incorrect
polarity)

Polarity reversed (0 - 1)

I'll check it when I can get physical access. Does anyone know if hangup
detection is disabled if the driver thinks the line polarity is
incorrect?


 
  Any ideas?
 
 Well, you can start by telling us the version of asterisk you're running, and 
 the date of the CVS pull if it's from CVS.  You could also post your 
 zapata.conf and zaptel.conf files. 


Sorry... 

Ver is Asterisk stable 1.07

zapata.conf

;
; Zapata telephony interface
;
; Configuration file

[trunkgroups]
;
; Trunk groups are used for NFAS or GR-303 connections.
;
; Group: Defines a trunk group.  
;group = trunkgroup,dchannel[,backup1...]
;
;trunkgroup  is the numerical trunk group to create
;dchannelis the zap channel which will have the 
;d-channel for the trunk.
;backup1 is an optional list of backup d-channels.
;
;trunkgroup = 1,24,48
;
; Spanmap: Associates a span with a trunk group
;spanmap = zapspan,trunkgroup[,logicalspan]
;
;zapspan is the zap span number to associate
;trunkgroup  is the trunkgroup (specified above) for the mapping
;logicalspan is the logical span number within the trunk group
to use.
;if unspecified, no logical span number is used.
;
;spanmap = 1,1,1
;spanmap = 2,1,2
;spanmap = 3,1,3
;spanmap = 4,1,4

[channels]
;
; Default language
;
;language=en
;
; Default context
;
context=default
;
; Switchtype:  Only used for PRI.
;
; national:   National ISDN 2 (default)
; dms100: Nortel DMS100
; 4ess:   ATT 4ESS
; 5ess:   Lucent 5ESS
; euroisdn:   EuroISDN
; ni1:Old National ISDN 1
;
switchtype=national
;
; Some switches (ATT especially) require network specific facility IE
; supported values are currently 'none', 'sdn', 'megacom', 'accunet'
;
;nsf=none
;
; PRI Dialplan:  Only RARELY used for PRI.
;
; unknown:Unknown
; private:Private ISDN
; local:  Local ISDN
; national:   National ISDN
; international:  International ISDN
;
;pridialplan=national
;
; PRI Local Dialplan:  Only RARELY used for PRI (sets the calling
number's numbering plan)
;
; unknown:Unknown
; private:Private ISDN
; local:  Local ISDN
; national:   National ISDN
; international:  International ISDN
;
;prilocaldialplan=national
;
; PRI callerid prefixes based on the given TON/NPI (dialplan)
; This is especially needed for euroisdn E1-PRIs
; 
; sample 1 for Germany 
;internationalprefix = 00
;nationalprefix = 0
;localprefix = 0711
;privateprefix = 07115678
;unknownprefix = 
;
; sample 2 for Germany 
;internationalprefix = +
;nationalprefix = +49
;localprefix = +49711
;privateprefix = +497115678
;unknownprefix = 
;
; PRI resetinterval: sets the time in seconds between restart of unused
channels, defaults to 3600
; minimum 60 seconds
; some PBXs don't like channel restarts. so set the interval to a very
long interval e.g. 1
;
;resetinterval = 3600 
;
; Overlap dialing mode (sending overlap digits)
;
;overlapdial=yes
;
; PRI Out of band indications.
; Enable this to report Busy and Congestion on a PRI using out-of-band
; notification. Inband indication, as used by Asterisk doesn't seem to
work
; with all telcos.
; 
; outofband:  Signal Busy/Congestion out of band with
RELEASE/DISCONNECT
; inband: Signal Busy/Congestion using in-band tones
;
; priindication = outofband
;
; ISDN Timers
; All of the ISDN timers and counters that are used are configurable.
Specify 
; the timer name, and its value (in ms for timers)
;
; pritimer = t200,1000
; pritimer = t313,4000
;
;
; Signalling method (default is fxs).  Valid values:
; em:  E  M
; em_w:E  M Wink
; featd:   Feature Group D (The fake, Adtran style, DTMF)
; featdmf: Feature Group D (The real thing, MF (domestic, US))
; featb:   Feature Group B (MF (domestic, US))
; fxs_ls:  FXS (Loop Start)
; fxs_gs:  FXS (Ground Start)
; fxs_ks:  FXS (Kewl Start)
; fxo_ls:  FXO (Loop Start)
; fxo_gs:  FXO (Ground Start)
; fxo_ks:  FXO (Kewl Start)
; pri_cpe: PRI signalling, CPE side
; pri_net: PRI signalling, Network side
; gr303fxoks_net: GR-303 Signalling, FXO Loopstart, Network side
; gr303fxsks_cpe: GR-303 Signalling, FXS Loopstart, CPE side
; sf: SF (Inband Tone) 

[Asterisk-Users] CVS Head, Flex 2.5.31 or higher? READ THIS!

[Steve Murphy]

Everyone using CVS head, and owning flex-2.5.31 (or higher)--

Please note that a new version of the expression ( $[  ]  constructs used in 
extensions.conf ) parser
is automatically built by the makefile if your flex is at 2.5.31 or higher. You 
can see what your
flex version is by saying flex -V...

Now, the new scanner has some nice things about it, but it does behave a little 
differently
in some (hopefully rare) situations. To help find one of these situations 
quickly, I submitted
a program for consideration to be included somewhere in the asterisk CVS, which 
checks for the
one problem I've seen reported so far. I advise everyone using cvs head and 
having flex-2.5.31, to
run this program the first time you build asterisk. You won't need it 
afterwards. Look at the 
expressions it flags, and if you don't understand the issue, read the 
README.variables file.

First, you can obtain the check_expr.c file by browsing to: 
http://bugs.digium.com/view.php?id=4491
and clicking on the attached file link.

Compile it with gcc -o check_expr -g check_expr.c

run it with ./check_expr /etc/asterisk/extensions.conf  (or whatever the 
path is on your machine).

an expr2_log file will be created, and every $[ ] expression it finds will be 
listed with its status.

If you get any warnings, look over the situation, and see if you have to do 
anything. The most likely
thing you might want to do is wrap some things in double quotes to keep the 
indicated operator from
being evaluated, such as regex expressions for the : operator.

Any questions, just write me. I'll do the best I can.

murf
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Re: [Asterisk-Users] CallerID/chan_sccp

[Julien Goodwin]

On 8/06/2005 11:37 PM, Sergio Chersovani wrote:

Joseph ha scritto:

When sending a call to a line defined on chan_sccp, there is an 
error on the console that says:


Jun  7 08:22:29 WARNING[3924]: sccp_channel.c:79
sccp_channel_send_callinfo: Incoming call SCCP/Line1-0008 doesn't
have CallerId name
  


Fixed, you can find the patch here
http://www.c-net.it/chan_sccp/
And this has been committed, should work through in about 5 hours 
(thanks sourceforge)

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[Asterisk-Users] Remote CDR logging on mysql:

[Tim Connolly]

I'm trying to setup 
remote CDR logging, as directed by:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20cdr%20odbc

Anyone have example 
of what I need to change to make an asterisk server log on a remote mysql 
server?



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Re: [Asterisk-Users] How to handle one incoming call on multiple lines?

[Henry Coleman]

Will do ..Thanks Henry

Andrew Kohlsmith wrote:


On Wednesday 08 June 2005 11:24, Henry Coleman wrote:
 


This feature is called  attendant - night answer position. Is it not
possible to switch the incoming call to an alternate extension based on
time of day ?
   



You need to read up.  This exact situation is given in the Asterisk Handbook.

http://www.digium.com/handbook-draft.pdf

In particular, you want GotoIfTime().

-A.
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Re: [Asterisk-Users] Remote CDR logging on mysql:

[Matthew Boehm]

Tim  wrote:

I'm trying to setup remote CDR logging, as directed by:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20cdr%20odbc
 
Anyone have example of what I need to change to make an asterisk server 
log on a remote mysql server?


If you are going to store CDRs on MySQL, why not skip ODBC and use the 
native way?


http://www.voip-info.org/wiki-Asterisk+cdr+mysql

We do and works great

-Matthew

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Re: [Asterisk-Users] TDM400P... ignoring hanguponpolarityswitch

[Andrew Kohlsmith]

On Wednesday 08 June 2005 12:00, Neil and Fiona wrote:
 /var/log/messages seems to be indicating that the wctdm driver thinks
 that the polarity of the line is reversed on start. (ie incorrect
 polarity)

 Polarity reversed (0 - 1)

Reverse the tip and ring on the line then.  :-)

 I'll check it when I can get physical access. Does anyone know if hangup
 detection is disabled if the driver thinks the line polarity is
 incorrect?

Could be, it's trivial to flip the tip and ring by accident.

 Sorry...
 Ver is Asterisk stable 1.07

Does stable have this feature?  I know HEAD does.  Do you see any mention of 
that configuration parameter in asterisk/channels/chan_zap.c?

 switchtype=national

Are you on a PRI?  This configuration option is meaningless if not.

 rxgain=11
 txgain=1

Have you actually tuned the system to get these parameters?  I'd use

http://lists.digium.com/pipermail/asterisk-users/2004-November/071301.html

to set these values.

 callerid=555

I'd probably use asreceived but this isn't causing any trouble.

 busydetect=yes

TURN THIS OFF!! unless you have a real reason to use it, do NOT use this... 
it's the #1 source of false hangups and other general weirdness.

It looks more or less fine, but yes, if it's saying it thinks the polarity's 
reversed from the get-go, I'd try to swap the tip and ring and see if that 
fixes things right.

-A.
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Re: [Asterisk-Users] Queue Log

[Hugo Begglo]

Thanks Johann.  - that helps out .

Johann wrote:


Hugo,

1118098465|1118098465.47|salesq|NONE|ENTERQUEUE||Ray Balbin 25 
(716)250-3405



1st column is unixtime stamp for the current date
2nd column is not really sure...maybe the duration?
3rd column is the queue name
4th column is their agent number (or NONE if there isn't one)

--johann
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Re: [Asterisk-Users] TDM400P... ignoring hanguponpolarityswitch

[Julian J. M.]

I've used that feature in asterisk HEAD, and it has worked for me (i
needed to apply a little patch for it to work for incoming calls
also), but i also used answeronpolarityswitch=yes. Maybe it's a logic
bug in the code. Try with that option and tell us the results ;)

BTW, it doesn't matter is the module detects the idle polarity as 1 or
-1... The code only checks for the polarity switch event (-11 or
1-1)

Julian.

On 6/8/05, Neil and Fiona [EMAIL PROTECTED] wrote:
 I've just had polarity reversal provisioned by our telco to test hangup
 detect with a TDM400P
 
 I've set hanguponpolarityswitch=yes in zapata.conf
 
 When I start Asterisk I get ignoring hanguponpolarityswitch
 in /var/log/asterisk/messages
 
 I assume that the option is either not valid or conflicts with another
 setting somewhere.
 
 Any ideas?
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Re: [Asterisk-Users] CallerID/chan_sccp

[Joseph]

On Thu, 2005-06-09 at 02:24 +1000, Julien Goodwin wrote:
 On 8/06/2005 11:37 PM, Sergio Chersovani wrote:
  Joseph ha scritto:
  
  When sending a call to a line defined on chan_sccp, there is an 
  error on the console that says:
 
  Jun  7 08:22:29 WARNING[3924]: sccp_channel.c:79
  sccp_channel_send_callinfo: Incoming call SCCP/Line1-0008 doesn't
  have CallerId name

  
  Fixed, you can find the patch here
  http://www.c-net.it/chan_sccp/
 And this has been committed, should work through in about 5 hours 
 (thanks sourceforge)

It works.

Thanks.

-- 
respectfully, Joseph ===
-= **  =

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[Asterisk-Users] Echo problem

[Martin Roy]

Ok I tried Digium TDM400 cards, I tried X100p cards, I tried Clipcomm  
CG-410 4 FXOs device. Now I just ordered a few Sipura 3000.


With the Digium TDM04B cards (4 FXOs) and X100p cards I tried the  
following :


echocancel=yes
echocancelwhenbridged=yes
echotraining=yes (I tried 800 with TDM04B cards but it didn't made  
any difference)

rxgain= I tried from -8.0 to 10.0
txgain = I tried from -8.0 to 10.0

by the way I live in Canada and the provider is Bell Canada for all  
lines (I have over 10 lines at one place and 3 lines at another places)


I tried on a bunch of different computers. I tried on a P4, a dual  
Xeon, a dual AMD Opteron, a bunch of Macs too (for X100p cards) a  
PowerMac 8500, 9600, 9650, G3 Desktop, G3 BW and G4.


I have echo problem on all of them. I even tried on different OS.  
Fedora Core 1, 2 and 3 for the PCs and Yellow Dog Linux 3.01 and 4.0  
for the Macs. I even tried the Clipcomm CG-410 on OS X (10.2.8,  
10.3.x and even 10.4)


I'm using Cisco IP Phone 7960 with SIP firmware 7.3 and a few  
Granstream GXP-2000. The echo is a lot worst on Cisco phones.


Now I just ordered 5 Sipura 3000 to see if that will remove the echo.  
I can't understand why it wouldn't work with the Digium cards...


If someone has a clue to help me figure out how to remove this echo  
well let me know as right now I'm considering that all Digium cards  
sucks... For Clipcomm well the echo was there and I can't get Caller  
ID to work so it's useless...


Thanks

Martin
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RE: [Asterisk-Users] Help Connecting Cisco AS5300 to Asterisk

[Leandro Tenorio]

The configuration in the blog does not depend on the product, it
depend on the IOS used. Should work for your 5300, the only problem you
could have, AFAIR is with the SIP-ua config. Authentication, starts after
12.2.something.
If you have problem come back and I give u a workaround.

LTenorio

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of chawki hammoud
Sent: Wednesday, June 08, 2005 6:36 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Help Connecting Cisco AS5300 to Asterisk


I have already looked into this page.  I thought this was for AS 5350, I am
not familiar with Cisco products and I don't know if there is a difference.
And there is no Asterisk set-up in this example.

Regards;

--- Stefan Reuter [EMAIL PROTECTED] wrote:

  It doesn't have to be IAX. Do you know how to configure it with 
  another protocol?
 
 have a look at

http://ertw.com/blog/archives/asterisk_and_an_as5350_sip_peer-190405.html
 
 =Stefan
 
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RE: [Asterisk-Users] Help Connecting Cisco AS5300 to Asterisk

[Leandro Tenorio]

Yes you can. There are some examples @ cisco look for TDM switching.

LTenorio

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Marcelo
Pacheco
Sent: Wednesday, June 08, 2005 9:52 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Help Connecting Cisco AS5300 to Asterisk

Stefan,

Is it possible to have the Cisco forward calls between T1 or E1 interefaces,
without VOIP DSPs, but only Modem DSPs ?

I need to have an AS5350 that is currently configured as a dial-in RAS to
forward incoming calls to Asterisk, but I can't do it with SIP, as I don't
have VOIP DSPs on the AS5350, only modem DSPs.

TIA,

Marcelo Pacheco

Em Qua 08 Jun 2005 06:36, chawki hammoud escreveu:
 I have already looked into this page.  I thought this was for AS 5350, 
 I am not familiar with Cisco products and I don't know if there is a 
 difference. And there is no Asterisk set-up in this example.

 Regards;

 --- Stefan Reuter [EMAIL PROTECTED] wrote:
   It doesn't have to be IAX. Do you know how to configure it with 
   another protocol?
 
  have a look at

 http://ertw.com/blog/archives/asterisk_and_an_as5350_sip_peer-190405.h
 tml

  =Stefan
 
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Re: [Asterisk-Users] Echo problem

[Michael D Schelin]

Hi Martin, There was an great post last week about echo. It stated that 
the order of the lines matters. It does. The channels must be listed 
last for the echo cancel and most other things to work. Rx and TX gain 
is one of the things also affected. Now I'm using TE110 card in my 
system. I hope this helps because I'm not sure about Analog lines.






Martin Roy wrote:

Ok I tried Digium TDM400 cards, I tried X100p cards, I tried Clipcomm  
CG-410 4 FXOs device. Now I just ordered a few Sipura 3000.


With the Digium TDM04B cards (4 FXOs) and X100p cards I tried the  
following :


echocancel=yes
echocancelwhenbridged=yes
echotraining=yes (I tried 800 with TDM04B cards but it didn't made  any 
difference)

rxgain= I tried from -8.0 to 10.0
txgain = I tried from -8.0 to 10.0

by the way I live in Canada and the provider is Bell Canada for all  
lines (I have over 10 lines at one place and 3 lines at another places)


I tried on a bunch of different computers. I tried on a P4, a dual  
Xeon, a dual AMD Opteron, a bunch of Macs too (for X100p cards) a  
PowerMac 8500, 9600, 9650, G3 Desktop, G3 BW and G4.


I have echo problem on all of them. I even tried on different OS.  
Fedora Core 1, 2 and 3 for the PCs and Yellow Dog Linux 3.01 and 4.0  
for the Macs. I even tried the Clipcomm CG-410 on OS X (10.2.8,  10.3.x 
and even 10.4)


I'm using Cisco IP Phone 7960 with SIP firmware 7.3 and a few  
Granstream GXP-2000. The echo is a lot worst on Cisco phones.


Now I just ordered 5 Sipura 3000 to see if that will remove the echo.  I 
can't understand why it wouldn't work with the Digium cards...


If someone has a clue to help me figure out how to remove this echo  
well let me know as right now I'm considering that all Digium cards  
sucks... For Clipcomm well the echo was there and I can't get Caller  ID 
to work so it's useless...


Thanks

Martin
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[Asterisk-Users] Number of AGI's running at the same time

[Jerry Geis]

Is there any metric on the number of AGI's that can run
at the
same time. Shouldnt be a limit in my mind but I am thinking in
terms of system performance.

My AGI is a C program with 3 meg executable size. 

Thanks,

Jerry




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[Asterisk-Users] Asterisk and Alcatel 4200 PBX

[=?ISO-8859-1?Q?Jos=E9?= Luis =?ISO-8859-1?Q?G=F3mez?=]

Hello list.
I'm going te explain my trouble.
I have my asterisk with a TDM400P with 4 FXS channels. Two ports are
connected to a Panasonic PBX (it's working fine), and others two ports
are connected to an Alcatel 4200 PBX (but it doesn't anwer). I connected
to a CO port (where i had a pstn line).
When I call to the Alcatel PBX, the asterisk show me in it console that
es ringing but never anwer.
I had configured with diferent signalling:
1-
zaptel.conf -  fxoks=3,4
zapata.conf -  signalling=fxo_ks
2-
zaptel.conf -  fxols=3,4
zapata.conf -  signalling=fxo_ls
3-
zaptel.conf -  fxogs=3,4
zapata.conf -  signalling=fxo_gs

I put 3,4 because 1,2 are connected to Panasonic PBX (working with ks).
But the PBX never anwer.
I also change the zone (because the PBX es french):
zaptel.conf -  loadzone=fr
defaultzone=fr
But it doesn't work.

My files:
* zaptel.conf
loadzone=us
defaultzone=us
fxoks=1-4

* zapata.conf
[channels]
busydetect=yes
busycount=10
cancallforward=yes
callprogress=no
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes 
immediate=no
signalling=fxo_ks 
rxgain=0.0
txgain=0.0
transfer=yes  
usecallerid=no

context=remotas1
channel = 1
context=remotas2
channel = 2
context=remotas3
channel = 3
context=remotas4
channel = 4

I don't know what to try.
Please help me.
Thanks.
   José Luis Gómez

-- 

José Luis Gómez
Qualis Information Technology
Av. Rivadavia 2553, PB Of. 43 EP
0342-4565684 int 102
www.qualis.com.ar
Soporte 0810-8880022
Santa Fe - Argentina

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Re: [Asterisk-Users] Echo problem

[Andrew Kohlsmith]

On Wednesday 08 June 2005 13:37, Martin Roy wrote:
 rxgain= I tried from -8.0 to 10.0
 txgain = I tried from -8.0 to 10.0

Unless you are making measurements and actually analyzing the results you're 
only stabbing in the dark playing with these things.

 by the way I live in Canada and the provider is Bell Canada for all
 lines (I have over 10 lines at one place and 3 lines at another places)

Bell's usually pretty good (I'm a Bell customer too) so unless you've got 
seriously screwey lines (unbalanced, reversed tip/ring, grounding issues) you 
should not be having this kind of problem.

Take a read here.  I reference this document continuously:

http://lists.digium.com/pipermail/asterisk-users/2004-November/071301.html

Yes, it's work and yes, you may have some trouble doing it/locating the 
numbers for milliwatt and quiet term but you know what, this is engineering 
and this is how to do it correctly.  Everything else is just pissing around 
hoping for a solution rather than making educated guesses and anlyzing the 
results.

 I tried on a bunch of different computers. I tried on a P4, a dual
 Xeon, a dual AMD Opteron, a bunch of Macs too (for X100p cards) a
 PowerMac 8500, 9600, 9650, G3 Desktop, G3 BW and G4.

 I have echo problem on all of them. I even tried on different OS.
 Fedora Core 1, 2 and 3 for the PCs and Yellow Dog Linux 3.01 and 4.0
 for the Macs. I even tried the Clipcomm CG-410 on OS X (10.2.8,
 10.3.x and even 10.4)

You're just stabbing in the dark here.

 I'm using Cisco IP Phone 7960 with SIP firmware 7.3 and a few
 Granstream GXP-2000. The echo is a lot worst on Cisco phones.

Interesting.

 Now I just ordered 5 Sipura 3000 to see if that will remove the echo.
 I can't understand why it wouldn't work with the Digium cards...

 If someone has a clue to help me figure out how to remove this echo
 well let me know as right now I'm considering that all Digium cards
 sucks... For Clipcomm well the echo was there and I can't get Caller
 ID to work so it's useless...

Follow the instructions on the link provided.  Find the milliwatt and quiet 
term numbers for your local CO.  Corner a Bell tech (most of them are really 
really good guys) and explain that you're trying to interface to a telephone 
line with your computer and you need the quiet and milliwatt numbers in order 
to ensure your gains are set correctly.  It's hidden info but not secret 
info.

Make sure your tip and ring aren't reversed.  Make sure one's not grounded or 
that there's not something else squirrely with your lines.

There is a (simple) FIR filter available on the TDM400P FXO modules.  Use the 
fxotune util to properly adjust it.

Echo is able to be eliminated, it's just sometimes a real tricky bugger to 
track down the cause.

Regards,
Andrew
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[Asterisk-Users] rxfax not working

[Jay Austad]

I have asterisk 1.0.7 and I made the required patch and got everything 
installed.  I have libtiff 3.7.0, and I'm using the zaptel stuff.

When I send a fax to it, it autodetects the fax and starts rxfax, however, the 
fax machine just sits at 1% and then disconnects.  I don't have any error 
messages or anything.  Any ideas as to why this isn't working or any 
suggestions for troubleshooting this further?

~jay
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Re: [Asterisk-Users] Queue Log

[Brian Roy]

On 6/7/05, Johann [EMAIL PROTECTED] wrote:
 Hugo,
 
  1118098465|1118098465.47|salesq|NONE|ENTERQUEUE||Ray Balbin 25
  (716)250-3405

 2nd column is not really sure...maybe the duration?

Asterisk UniqueID of the call. 

-Brian
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RE: [Asterisk-Users] * @ Home: All Circuits busy

[[EMAIL PROTECTED]]

did genzaptelconf -s -d say it found any cards? 

--- [EMAIL PROTECTED] wrote:

 Dean,
 
 Actually, I have run genzaptelconf -s -d but it
 still didn’t seem to modify any 
 of the config files that I look at in the AMP
 console.  Should I try modifying 
 the config files manually?
 Thanks,
 Marc
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:asterisk-users-
 [EMAIL PROTECTED]  On Behalf Of Dean Mumby
 Sent: Wednesday, June 08, 2005 10:07 AM
 To:   Asterisk Users Mailing List - Non-Commercial
 Discussion
 Subject:  Re: [Asterisk-Users] * @ Home: All Circuits
 busy
 
 [EMAIL PROTECTED] wrote:
 All,
 
 I have an [EMAIL PROTECTED] installation with a TDM40B
 card.  I can make internal 
 IP calls with no problems, but when I try to dial
 out I get a message that “All 
 Circuits are Busy”.  I looked into the Zapata.conf
 files and such but see no 
 modifications.  Is there a step that I am missing?? 
 Does anyone have 
 documentation of step-by-step config for this TDM40B
 card?
 
 Thanks,
 Marc 
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 have you run the genzaptelconf -s
 try aah-help for more info
 
 
 Dean
 
 
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RE: [Asterisk-Users] Echo problem

[Jon Califf]

I use Digium TDM400 cards as well. Asterisk's software echo cancellation
sucks. From what I've heard on the IRC channel, you'll never completely
eliminate echo with it. And unfortunately, hardware echo cancellation starts
out at a full T1. They don't seem to have any solution for someone with 4
pots lines like myself. 

I haven't been able to completely eliminate echo, but I've come close by
using the following:

echocancel=64
echocancelwhenbridged=no
echotraining=800
rxgain=4.5
txgain=0.0

echocancel=64 was significantly better than echocancel=128 (supposedly the
same setting you get when you use echocancel=yes)

echotraining at 400 was too short, but 800 seems to almost completely
eliminate any initial echo. Occasionally there is still a little echo to
start with, but it isn't very bad and it goes away quickly.

What sort of echo are you getting? Loud, quiet, fades in and out, starts
halfway through the call, starts loud and gets quit?

Jon


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Martin Roy
Sent: Wednesday, June 08, 2005 10:37 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Echo problem

Ok I tried Digium TDM400 cards, I tried X100p cards, I tried Clipcomm  
CG-410 4 FXOs device. Now I just ordered a few Sipura 3000.

With the Digium TDM04B cards (4 FXOs) and X100p cards I tried the  
following :

echocancel=yes
echocancelwhenbridged=yes
echotraining=yes (I tried 800 with TDM04B cards but it didn't made  
any difference)
rxgain= I tried from -8.0 to 10.0
txgain = I tried from -8.0 to 10.0

by the way I live in Canada and the provider is Bell Canada for all  
lines (I have over 10 lines at one place and 3 lines at another places)

I tried on a bunch of different computers. I tried on a P4, a dual  
Xeon, a dual AMD Opteron, a bunch of Macs too (for X100p cards) a  
PowerMac 8500, 9600, 9650, G3 Desktop, G3 BW and G4.

I have echo problem on all of them. I even tried on different OS.  
Fedora Core 1, 2 and 3 for the PCs and Yellow Dog Linux 3.01 and 4.0  
for the Macs. I even tried the Clipcomm CG-410 on OS X (10.2.8,  
10.3.x and even 10.4)

I'm using Cisco IP Phone 7960 with SIP firmware 7.3 and a few  
Granstream GXP-2000. The echo is a lot worst on Cisco phones.

Now I just ordered 5 Sipura 3000 to see if that will remove the echo.  
I can't understand why it wouldn't work with the Digium cards...

If someone has a clue to help me figure out how to remove this echo  
well let me know as right now I'm considering that all Digium cards  
sucks... For Clipcomm well the echo was there and I can't get Caller  
ID to work so it's useless...

Thanks

Martin
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Re: [Asterisk-Users] * @ Home: All Circuits busy

[Greg Jones Media]

I have seen the same problem.  The zaptel hardware looks fine in zttool and
appears to be ok when genzaptel -s -d is run, but when you look at the zap
channels in CLI, you only see the pseudo channel.


- Original Message - 
From: [EMAIL PROTECTED]

To: asterisk-users@lists.digium.com
Sent: Wednesday, June 08, 2005 11:45 AM
Subject: RE: [Asterisk-Users] * @ Home: All Circuits busy



Dean,

Actually, I have run genzaptelconf -s -d but it still didn't seem to
modify any
of the config files that I look at in the AMP console.  Should I try
modifying
the config files manually?
Thanks,
Marc
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED]  On Behalf Of Dean Mumby
Sent: Wednesday, June 08, 2005 10:07 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] * @ Home: All Circuits busy

[EMAIL PROTECTED] wrote:
All,

I have an [EMAIL PROTECTED] installation with a TDM40B card.  I can make
internal
IP calls with no problems, but when I try to dial out I get a message that
All
Circuits are Busy.  I looked into the Zapata.conf files and such but see
no
modifications.  Is there a step that I am missing??  Does anyone have
documentation of step-by-step config for this TDM40B card?

Thanks,
Marc
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have you run the genzaptelconf -s
try aah-help for more info


Dean


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Re: [Asterisk-Users] Multiple E1s on one box

[Franco Bellagamba]

Jorge,

As far as I've read, you won't be able to handle 8 E1 in one box.

By the way, have you had success with interconnecting E1 R2 argentina? I´m
having trouble with a Meridian... I can only make calls from asterisk, but
the other way arround...

Tks
Franco
- Original Message - 
From: Jorge Alayon [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, June 07, 2005 11:13 AM
Subject: [Asterisk-Users] Multiple E1s on one box


 Hello all,

 Has anyone tried 8xE1 in one box using Asterisk and Digium boards ?
 What is the CPU needed for sustained performance in this capacity ?
 Is this affected if G.729 codec is used ?


 Regards,

 Jorge A.
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RE: [Asterisk-Users] * @ Home: All Circuits busy

[maoleson]

Dean,

Here are the results of the genzaptelconf -s -d.  As you can see, it is 
throwing some errors, but I am a bit of a newbie so any help you could provide 
would be greatly appreciated!

[EMAIL PROTECTED] /]# genzaptelconf -s -d


STOPPING ASTERISK
Asterisk ended with exit status 0
Asterisk shutdown normally.

Disconnected from Asterisk server
Asterisk Stopped

STOPPING FOP SERVER
FOP Server Stopped
Hint: insmod errors can be caused by incorrect module parameters, including 
invalid IO or IRQ parameters.
  You may find more information in syslog or the output from dmesg
Hint: insmod errors can be caused by incorrect module parameters, including 
invalid IO or IRQ parameters.
  You may find more information in syslog or the output from dmesg
Hint: insmod errors can be caused by incorrect module parameters, including 
invalid IO or IRQ parameters.
  You may find more information in syslog or the output from dmesg
Hint: insmod errors can be caused by incorrect module parameters, including 
invalid IO or IRQ parameters.
  You may find more information in syslog or the output from dmesg
Hint: insmod errors can be caused by incorrect module parameters, including 
invalid IO or IRQ parameters.
  You may find more information in syslog or the output from dmesg
Hint: insmod errors can be caused by incorrect module parameters, including 
invalid IO or IRQ parameters.
  You may find more information in syslog or the output from dmesg
Hint: insmod errors can be caused by incorrect module parameters, including 
invalid IO or IRQ parameters.
  You may find more information in syslog or the output from dmesg
Hint: insmod errors can be caused by incorrect module parameters, including 
invalid IO or IRQ parameters.
  You may find more information in syslog or the output from dmesg
Unloading zaptel hardware drivers:
Removing zaptel module:  zaptel: Device or resource busy
   [FAILED]
Loading zaptel framework:  [  OK  ]
Waiting for zap to come online ...OK
Loading zaptel hardware modules:
Running ztcfg: [  OK  ]

SETTING FILE PERMISSIONS
Permissions OK

STARTING ASTERISK
Asterisk Started

STARTING FOP SERVER
FOP Server Started

** SIP/200 in position 2
** SIP/201 in position 3
** SIP/202 in position 4
   Chan Extension  Context Language   MusicOnHold
 pseudofrom-pstn   en
Verbosity is at least 3
[EMAIL PROTECTED] /]#
[EMAIL PROTECTED] /]#


Thanks,
Marc
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